Method for recovering audio signals, terminal and storage medium

ABSTRACT

The present disclosure, belonging to the field of audio technology, provides a method and apparatus for recovering audio signals. The method includes: buffering an audio signal sampled at a preset number of sampling points each time, and then performing frequency spectrum analysis on the sampled audio signal by FFT; when it is determined that the audio signal is compressed, filtering a frequency point; recovering high-frequency signals based on audio signals before the frequency point; and performing phase recovery on the high-frequency signals. The present disclosure provides the method for recovering audio signals.

CROSS-REFERENCE TO RELATED APPLICATION

This application is a national phase of PCT patent application No.:PCT/CN2018/117766 filed on Nov. 27, 2018, which claims priority toChinese Patent Application No. 201811053050.0, filed on Sep. 10, 2018and entitled “METHOD AND APPARATUS FOR RECOVERING AUDIO SIGNALS”, theentire contents of which are incorporated herein by reference.

TECHNICAL FIELD

The present disclosure relates to the field of audio technology, andmore particularly, relates to a method for recovering audio signals, aterminal and a storage medium.

BACKGROUND

In the audio field, in order to save audio data transmission resources,audio data is generally subjected to low-pass filtering first to filterhigh-frequency signals that are insensitive to the human auditorysystem, and the audio data subjected to low-pass filtering is thencompressed to increase the compression ratio and reduce the amount ofaudio data.

SUMMARY

Embodiments of the present disclosure provide a method for recoveringaudio signals, a terminal and a storage medium.

In a first aspect, a method for recovering audio signals is provided.The method includes:

determining a first frequency point in an audio signal having ahigh-frequency signal to be recovered by power spectrum scanning,wherein the first frequency point is a frequency point having a minimumfrequency of the high-frequency signal to be recovered in the audiosignal;

among a plurality of frequency subbands having equal width of the audiosignal having the high-frequency signal to be recovered, recovering,according to the audio signal of a previous frequency subband of atarget frequency subband, the audio signal of the target frequencysubband and the audio signals of the frequency subbands after the targetfrequency subband in the plurality of frequency subbands, wherein thetarget frequency subband is a frequency subband to which the firstfrequency point belongs;

synthesizing the audio signals of the frequency subbands before thetarget frequency subband in the plurality of frequency subbands, theaudio signal of the target frequency subband, and the audio signals ofthe frequency subbands after the target frequency subband in theplurality of frequency subbands;

separating the synthesized audio signal according to the first frequencypoint to obtain high-frequency signals and low-frequency signals, andperforming phase recovery on the high-frequency signals; and

superimposing the high-frequency signals subjected to phase recovery andthe low-frequency signals to obtain an audio signal in which thehigh-frequency signals are recovered.

Optionally, the method further includes:

if the first frequency point is not present in the audio signal having ahigh-frequency signal to be recovered, converting the audio signalhaving a high-frequency signal to be recovered into a plurality offrequency subbands having an equal width, and synthesizing the audiosignals of the plurality of frequency subbands;

separating the audio signal obtained by synthesizing the audio signalsof the plurality of frequency subbands according to a preset thirdfrequency point to obtain high-frequency signals and low-frequencysignals; and

superimposing the high-frequency signals and the low-frequency signalsobtained by separating according to the preset third frequency point toobtain the audio signal in which the high-frequency signals arerecovered.

Optionally, separating the synthesized audio signal according to thefirst frequency point to obtain high-frequency signals and low-frequencysignals includes:

performing linear high-pass filtering on the synthesized audio signal toobtain the high-frequency signals, and performing linear low-passfiltering on the synthesized audio signal to obtain the low-frequencysignals, wherein a frequency of each of the signals subjected to linearhigh-pass filtering is greater than or equal to the frequency of thefirst frequency point, and a frequency of each of the signals subjectedto linear low-pass filtering is less than the frequency of the firstfrequency point.

Optionally, performing phase recovery on the high-frequency signalsincludes:

performing all-pass biquad infinite impulse response (IIR) filtering onthe high-frequency signals to obtain the high-frequency signalssubjected to phase recovery.

Optionally, the method further includes:

determining a coefficient of the biquad IIR filtering according to thefrequency of the first frequency point and sampling rates.

Optionally, determining a first frequency point in an audio signalhaving a high-frequency signal to be recovered by power spectrumscanning includes:

buffering an audio signal which is sampled at a preset number ofsampling points;

performing fast Fourier transform (FFT) on the sampled audio signal toobtain an FFT result;

according to the FFT result, finding a first frequency point thatsatisfying preset conditions, wherein the preset conditions are that adifference between frequencies of the first frequency point and a secondfrequency point is less than a first preset value, a difference betweenpowers of the first frequency point and the second frequency point isgreater than a second preset value, a power of a frequency point havinga frequency greater than the frequency of the first frequency point iszero, and the frequency of the second frequency point is less than thefrequency of the first frequency point.

Optionally, prior to the performing FFT on the sampled audio signal toobtain an FFT result, the method further includes:

windowing the sampled audio signal to obtain audio signal subjected towindowing; and

wherein performing FFT on the sampled audio signal to obtain an FFTresult includes:

performing the FFT on the audio signal subjected to windowing to obtainthe FFT result.

In a second aspect, an apparatus for recovering audio signals isprovided. The apparatus includes:

a buffering module, configured to buffer an audio signal sampled at apreset number of sampling points;

a fast Fourier transform (FFT) module, configured to perform FFT on thesampled audio signal to obtain an FFT result;

a converting module, configured to, according to the FFT result, if afirst frequency point satisfying preset conditions is present, convertthe audio signal sampled at the preset number of sampling points intoaudio signals of a plurality of frequency subbands having an equalwidth;

a determining module, configured to determine a target frequency subbandto which the first frequency point belongs, wherein the presetconditions are that a difference between frequencies of the firstfrequency point and a second frequency point is less than a first presetvalue, a difference between powers of the first frequency point and thesecond frequency point is greater than a second preset value, a power ofa frequency point having a frequency greater than the frequency of thefirst frequency point is zero, and the frequency of the second frequencypoint is less than the frequency of the first frequency point;

a recovering module, configured to recover, according to the audiosignal of a previous frequency subband of the target frequency subband,the audio signal of the target frequency subband in the plurality offrequency subbands and the audio signals of the frequency subbands afterthe target frequency subband;

a synthesizing module, configured to synthesize the audio signals of thefrequency subbands before the target frequency subband in the pluralityof frequency subbands, the audio signal of the target frequency subband,and the audio signals of the frequency subbands after the targetfrequency subband in the plurality of frequency subbands;

a separating module, configured to separate the synthesized audio signalaccording to the first frequency point to obtain high-frequency signalsand low-frequency signals, wherein the recovery module is furtherconfigured to perform phase recovery on the high-frequency signals; and

a superimposing module, configured to superimpose the high-frequencysignals subjected to phase recovery and the low-frequency signalsobtained by separating to obtain sampled audio signal in which thehigh-frequency signals are recovered.

Optionally, the converting module is further configured to, according tothe FFT result, if the first frequency point is not present, convert theaudio signal sampled at the preset number of sampling points into aplurality of frequency subbands having an equal width;

the synthesizing module is further configured to synthesize the audiosignals of the plurality of frequency subbands;

the separating module is further configured to separate the audio signalobtained by synthesizing the audio signals of the plurality of frequencysubbands according to a preset third frequency point to obtainhigh-frequency signals and low-frequency signals; and

the superimposing module is further configured to superimpose thehigh-frequency signals and the low-frequency signals according to thepreset third frequency point to obtain the sampled audio signal.

Optionally, the separating module is configured to:

perform linear high-pass filtering on the synthesized audio signal toobtain the high-frequency signals, and perform linear low-pass filteringon the synthesized audio signal to obtain the low-frequency signals,wherein a frequency of each of the signals subjected to linear high-passfiltering is greater than or equal to the frequency of the firstfrequency point, and a frequency of each of the signals subjected tolinear low-pass filtering is less than the frequency of the firstfrequency point.

Optionally, the recovering module is configured to:

perform all-pass biquad IIR filtering on the high-frequency signals toobtain high-frequency signals subjected to phase recovery.

Optionally, the determining module is further configured to:

determine a coefficient of the biquad IIR filtering according to thefrequency of the first frequency point and sampling rates.

Optionally, the apparatus further includes:

a windowing module, configured to, prior to performing FFT on thesampled audio signal to obtain an FFT result, windowing the sampledaudio signal to obtain audio signal subjected to windowing; and

the FFT module is configured to:

perform the FFT on the audio signal subjected to windowing to obtain theFFT result.

In a third aspect, a terminal is provided. The terminal includes amemory and a processor, the memory is used to store at least oneinstruction, and the processor is used to implement any method accordingthe first aspect when executing the at least one instruction.

In a fourth aspect, a non-transitory computer-readable storage medium isprovided. The storage medium stores at least one instruction, and the atleast one instruction, when being executed by a processor, implementsany method according the first aspect.

The technical solutions according to the embodiments of the presentdisclosure at least achieve the following beneficial effects.

In the embodiments of the present disclosure, in case of an audio with alossy format, after an audio signal sampled at a preset number ofsampling points is buffered each time, the sampled audio signal may besubjected to FFT to obtain an FFT result. According to the FFT result,if a first frequency point satisfying preset conditions is present, theaudio signal sampled at the preset number of sampling points areconverted into audio signals of a plurality of frequency subbands havingan equal width. A target frequency subband including the first frequencypoint is determined. Then, based on the audio signal of a previousfrequency subband of the target frequency subband, the audio signal ofthe target frequency subband in the plurality of frequency subbands andthe audio signals of the frequency subbands after the target frequencysubband are recovered. Next, the audio signals of the frequency subbandsbefore the target frequency subband, the audio signal of the targetfrequency subband, and the audio signals of the frequency subbands afterthe target frequency subband in the plurality of frequency subbands aresynthesized. The synthesized audio signal are separated according to thefirst frequency point to obtain high-frequency signals and low-frequencysignals, and the high-frequency signals are subjected to phase recovery.The high-frequency signals subjected to phase recovery and thelow-frequency signals are superimposed to obtain sampled audio signal inwhich the high-frequency signals are recovered. As such, since thehigh-frequency signals in the sampled audio signal may be recovered, thesampled audio signal are recovered as well. Therefore, the method forrecovering audio signals is provided.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a flowchart of a method for recovering audio signals asprovided by an embodiment of the present disclosure;

FIG. 2 is a schematic diagram of filtered frequency points as providedby an embodiment of the present disclosure;

FIG. 3 is a schematic structural diagram of an apparatus for recoveringaudio signals as provided by an embodiment of the present disclosure;

FIG. 4 is a schematic structural diagram of an apparatus for recoveringaudio signals as provided by an embodiment of the present disclosure;and

FIG. 5 is a schematic structural diagram of a terminal as provided by anembodiment of the present disclosure.

DETAILED DESCRIPTION

The embodiments of the present disclosure will be described in furtherdetail with reference to the attached drawings, to clearly present theobjects, technical solutions, and advantages of the present disclosure.

The embodiments of the present disclosure provide a method forrecovering audio signals. An execution subject body of the method may bea terminal. The terminal may be a mobile phone, a computer, a tabletcomputer, or the like.

A processor, a memory, and a transceiver may be configured in theterminal. The processor may be configured to recover audio signals. Thememory may be configured to recover desired data and generated dataduring recovering the audio signals. The transceiver may be configuredto receive and transmit data. The terminal may further include aninput/output device such as a screen, wherein the screen may be a touchscreen. The screen may be configured to display recovered audio signals,and the like.

In the embodiments of the present disclosure, a mobile phone may be, forexample, used as the terminal for detailed description of practice ofthe technical solutions, and other cases are similar and may not berepeated again herein.

Prior to the practice, the application scenario of the embodiments ofthe present disclosure is first introduced:

In the audio field, in order to save audio data transmission resources,audio data is generally subjected to low-pass filtering first to filterhigh-frequency signals that are insensitive to the human auditorysystem, and the audio data subjected to low-pass filtering is thencompressed to increase the compression ratio and reduce the amount ofaudio data. With the development of computer technologies and theimprovement of quality of audio digital-to-analog converters andearphones, when the audio data is played, the defects caused by thefiltered high-frequency signals become more and more obvious. Therefore,a method for recovering high-frequency signals in the compressed audiosignals is desired.

An embodiment of the present disclosure provides a method for recoveringaudio signals. As shown in FIG. 1, the method may include the followingsteps.

In step 101, an audio signal sampled at a preset number of samplingpoints is buffered.

The preset number may be preset and stored in the terminal. The presetnumber generally ranges from 2048 to 32768, and is equal to 2^(N) (whichfacilitates the operation of subsequent FFT algorithm), where N isgreater than or equal to 11, and less than or equal to 15. For example,the preset number is 8192.

During the practice, after downloading a compressed audio, the terminalmay sample audio signals of the compressed audio according to a presetsampling rate. The audio signal sampled at a preset number of samplingpoints, which are buffered each time, is subjected to subsequentprocessing as a small block of audio signals.

It should be noted that, in the embodiment of the present disclosure,the longer the audio signal sampled by the sample points, which arebuffered each time, the higher the recovery quality. However, therequirements for hardware resources are relatively high, and therefore,the preset number should be selected appropriately, i.e., should besuitable for hardware resources and achieve a better recovery quality.

It should also be noted that the above sampling rate may be 22.05 KHz,44.1 KHz, or the like. The sampling method may be pulse code modulation(PCM) sampling.

In step 102, the sampled audio signal is subjected to fast Fouriertransform (FFT) to obtain an FFT result.

During the practice, upon obtaining a small block of audio signals, theterminal inputs the small block of audio signals into an FFT algorithmand perform FFT on the audio signals to obtain an FFT result. Forexample, when audio signal sampled by 8192 sample points (which may beconsidered as real-number sample points) are buffered, the obtained FFTresult has a length of (8192/2)+1=4097, that is, 4097 complex numbers.

It should be noted that the FFT is performed by using an real discreteFourier transform (RDFT) algorithm. The RDFT algorithm is a type of FFTand specifically used to sample real numbers in a time domain andconvert them into complex numbers in a frequency domain. After N realnumbers are subjected to RDFT, (N/2)+1 complex numbers will be obtained.Each complex number is subjected to a modulo operation, and (N/2)+1 realnumbers will be then obtained, which means the amplitudes of (N/2)+1frequency points. Each amplitude is calculated in log 10(X), where Xrepresents the amplitude, and a power spectrum is then obtained.

Optionally, prior to the FFT, the audio signals may also be subjected towindowing. The corresponding processing may be described as follows:

windowing the sampled audio signal to obtain audio signal subjected towindowing; and performing the FFT on the audio signal subjected towindowing to obtain the FFT result.

Windowing refers to multiplication of an original integrand and aspecific window function in Fourier integral. In consideration of thepassband flatness and the stopband attenuation, a NUTTALL window may beselected as a window function for windowing.

During the practice, the terminal may acquire a pre-stored windowfunction, window on the sampled audio signal by using the windowfunction to obtain audio signal subjected to windowing, then input theaudio signal subjected to windowing to FFT, and perform the FFT toobtain the FFT result.

It should be noted that the periodic extension is actually made in theFFT, this is because the data is processed by the terminal within alimited period of time. In the FFT, the desired time is an integral fromnegative infinity to positive infinity, and thus needs to be extended,and the problem of spectral leakage will be then involved. Therefore,the audio signals need to be subjected to windowing to correct theproblem of spectral leakage.

In step 103, according to the FFT result, if a first frequency pointsatisfying preset conditions is present, audio signal sampled at apreset number of sampling points are converted into audio signals of aplurality of frequency subbands having an equal width, and a targetfrequency subband to which the first frequency point belongs isdetermined.

As shown in FIG. 2, the preset conditions are that a difference betweenfrequencies of the first frequency point and a second frequency point isless than a first preset value, a difference between powers of the firstfrequency point and the second frequency point is greater than a secondpreset value, a power of a frequency point having a frequency greaterthan the frequency of the first frequency point is zero, and thefrequency of the second frequency point is less than the frequency ofthe first frequency point. The first preset value, such as 10 Hz, may bepreset and stored in the terminal. The second preset value, such as 6dB, may be preset and stored in the terminal.

During the practice, after obtaining the FFT result, if the FFT resultis a frequency spectrum, the terminal may calculate a power spectrum(which may be the square of an amplitude corresponding to each frequencypoint) according to the frequency spectrum. In the power spectrum, eachfrequency point corresponds to one power. The terminal may then scan thepower spectrum to find a cliff-like attenuation point of power, that is,to find a first frequency point satisfying the preset conditions. Thepreset conditions are that the frequency of the second frequency pointis less than the frequency of the first frequency point; the differencebetween the frequencies of the first frequency point and the secondfrequency point is less than the first preset value, the differencebetween the powers of the first frequency point and the second frequencypoint is greater than the second preset value; and a power of afrequency point having a frequency greater than the frequency of thefirst frequency point is zero. The first frequency point may be referredto as a cliff-like attenuation point.

After finding out the first frequency point, the terminal may acquirethe audio signal sampled at the preset number of sampling points in thepreset step 101, then window the audio signals by using a windowfunction (the window function may be a NUTTALL window function), and,after the windowing, convert the audio signal subjected to windowinginto audio signals of frequency subbands having an equal width by usinga preset modified discrete consine transform (MDCT) algorithm. Thefrequency subband in which the first frequency point is located issearched from these frequency subbands.

For example, the FFT result has a length of (8192/2)+1=4097, which maybe expressed as SPEC[0, 1 . . . , 4096]. Assuming that the firstfrequency point is N, the power difference is SPEC[N−1]-SPEC[N]≥a secondpreset value, and SPEC [N+1 . . . 4096] are all 0. The frequency of thefirst frequency point may be expressed as N*(4097/(sampling rate/2)) inHz. 4096 frequency subbands may be obtained through the MDCT algorithm,each frequency subband being equal in width. In addition, 4096 subbandsare equally divided (sampling rate/2) in Hz. The frequency subbands maybe named SUBBAND[0 . . . 4095]. It is assumed that the frequency subbandincluding the first frequency point is N, the frequency range of theSUBBAND[N] frequency subband includes the frequency of the firstfrequency point.

It should be noted that the method of obtaining the frequency subbandsby using the MDCT algorithm is merely an exemplary form, and frequencysubbands may also be obtained by using a polyphase filter.

It should also be noted that the first frequency point is actually afrequency point having the smallest frequency among the filteredfrequency points in the course of compression.

In step 104, according to the audio signal of a previous frequencysubband of the target frequency subband, the audio signal of the targetfrequency subband and the audio signals of the frequency subbands afterthe target frequency subband in the plurality of frequency subbands arerecovered.

During the practice, after the target frequency subband is found out,the previous frequency subband of the target frequency subband may bedetermined, the previous frequency subband being a frequency subbandhaving a frequency endpoint value less than a frequency endpoint valueof the target frequency subband and having the smallest difference fromthe frequency endpoint value of the target frequency subband. The audiosignal of the previous frequency subband is then acquired. The audiosignal of the target frequency subband and the audio signals of thefrequency subbands after the target frequency subband in the pluralityof frequency subbands are recovered.

The recovery process may be as follows:

It is assumed that the frequency subband containing the first frequencypoint is N, SUBBAND[K]=SUBBAND[K−1]*(SQRT(2)/2) may be used, whereN≤K≤4095, and SQRT indicating square root. It may be seen that the audiosignal of the first frequency subband isSUBBAND[N]=SUBBAND[N−1]*(SQRT(2)/2), and the audio signal of a frequencysubband next to the first frequency subband isSUBBAND[N+1]=SUBBAND[N]*(SQRT(2)/2). It may be seen that the audiosignal of the N^(th) frequency subband is determined by using the audiosignal of the (N−1)^(th) frequency subband, and the audio signal of the(N+1)^(th) frequency subband is determined by using the audio signal ofthe N^(th) frequency subband. The audio signal of the N^(th) frequencysubband and the audio signal of each of the subsequent frequencysubbands are calculated in turn. In this way, the audio signal of thetarget frequency subband and the audio signals of the frequency subbandsafter the target frequency subband may be recovered.

In step 105, the audio signals of the frequency subbands before thetarget frequency subband in the plurality of frequency subbands, theaudio signal of the target frequency subband, and the audio signals ofthe frequency subbands after the target frequency subband in theplurality of frequency subbands are synthesized.

During the practice, after recovering the audio signal of the targetfrequency subband and the audio signals of the audio subbands after thetarget frequency subband, the terminal may input the audio signals ofthe frequency subbands before the target frequency subband in theplurality of frequency subbands, the audio signal of the targetfrequency subband, and the audio signals of the frequency subbands afterthe target frequency subband in the plurality of frequency subbands toan inverse MDCT algorithm (since the frequency subbands are equallydivided by using the MDCT algorithm earlier, the inverse MDCT algorithmis used here) to obtain the synthesized audio signal, these synthesizedaudio signal including high-frequency signals.

In step 106, the synthesized audio signal is separated according to thefirst frequency point to obtain high-frequency signals and low-frequencysignals; and the high-frequency signals are subjected to phase recovery.

A frequency of each of the low-frequency signals is less than thefrequency of the first frequency point, and a frequency of each of thehigh-frequency signals is greater than or equal to the frequency of thefirst frequency point.

During the practice, the terminal may separate the synthesized audiosignal according to the first frequency point to obtain audio signals(which may be referred to as high-frequency signals) each having afrequency greater than the frequency of the first frequency point andaudio signals (which may be referred to as low-frequency signals) eachhaving a frequency less than the frequency of the first frequency point.

Since the audio signal of the N^(th) frequency subband is determined instep 105 by using the audio signal of the (N−1)^(th) frequency subband,the phase of the audio signal of the N^(th) frequency subband is thesame as the phase of the audio signal of the (N−1)^(th) frequencysubband, it is also necessary to correct the phases of thehigh-frequency signals. Therefore, the high-frequency signals may besubjected to phase recovery to obtain high-frequency signals subjectedto phase recovery.

Optionally, the high-frequency signals and the low-frequency signals maybe separated by a filter. The corresponding processing may be asfollows:

The synthesized audio signal are subjected to linear high-pass filteringto obtain high-frequency signals, and the synthesized audio signal aresubjected to linear low-pass filtering to obtain low-frequency signals.

A frequency of the signal subjected to linear high-pass filtering isgreater than or equal to the frequency of the first frequency point, anda frequency of the signal subjected to low-pass filtering is less thanthe frequency of the first frequency point.

During the practice, the terminal may input the synthesized audio signalinto a preset linear high-pass filtering algorithm, so that thehigh-frequency signals pass, and the low-frequency signals are filtered,thereby obtaining the high-frequency signals. In addition, thesynthesized audio signal may be input into a preset linear low-passfiltering algorithm, so that the low-frequency signals pass, and thehigh-frequency signals are filtered, thereby obtaining the low-frequencysignals.

It should be noted that the linear high-pass filtering algorithm and thelinear low-pass filtering algorithm may be an algorithm that implementsa function of a finite impulse response (FIR) linear filter and isdesigned by using a window function method, respectively. A NUTTALLwindow may be selected as a window function. The length may be oneeighth of the preset number in step 101 minus one.

In addition, when linear high-pass filtering is performed, the terminalmay be connected with a linear high-pass filter and a linear low-passfilter, and input the synthesized audio signal to the linear high-passfilter, such that the high-frequency signals pass, and the low-frequencysignals are filtered, thereby obtaining high-frequency signals, and thehigh-frequency signals are then returned to the terminal. In addition,the terminal may input the synthesized audio signal into a preset linearlow-pass filter, such that the low-frequency signals pass, and thehigh-frequency signals are filtered, thereby obtaining low-frequencysignals, and the low-frequency signals are then returned to theterminal.

It should be noted that the linear high-pass filter and the linearlow-pass filter may also be FIR linear filters designed by using awindow function method.

Optionally, the high-frequency signals are subjected to phase recoveryby using a filtering manner The corresponding processing may be asfollows:

the high-frequency signals are subjected to all-pass biquad infiniteimpulse response (IIR) filtering to obtain high-frequency signalssubjected to phase recovery.

During the practice, a common conductive wire transmits a groupextension characteristic of audio analog signals (i.e., the higher thefrequency of the audio signal, the larger the phase offset). Theterminal may input the high-frequency signals into an all-pass biquadIIR filtering algorithm. The all-pass biquad IIR filtering algorithm mayperform nonlinear phase offset on the high-frequency signals to obtainhigh-frequency signals subjected to phase recovery.

In addition, when performing phase recovery, the terminal may also beconnected with an all-pass biquad IIR filter, and transmit thehigh-frequency signals to the all-pass biquad IIR filter, such that thebiquad IIR filter performs nonlinear phase offset on the high-frequencysignals to obtain high-frequency signals subjected to phase recovery,and the high-frequency signals are then returned to the terminal.

Optionally, the all-pass biquad IIR filtering algorithm has differentcoefficients for different sampling rates. In the embodiment of thepresent disclosure, a process for determining the coefficients of theall-pass biquad IIR filtering algorithm (the coefficients may beconsidered as non-normalized coefficients) is also provided:

a coefficient of the biquad IIR filtering is determined according to thefrequency of the first frequency point and the sampling rates.

The non-normalized coefficients of the biquad IIR filtering algorithmare generally a0, a1, a2, b0, b1, b2. The frequency response curve andgain of the biquad IIR filtering algorithm may be determined accordingto these coefficients.

During the practice, in the calculation process, it may be firstcalculated:

G=tan(PI*(F/FS))   (1)

In the formula (1), tan represents a calculated tangent value; PIrepresents pi; F represents the frequency of the first frequency point;and FS represents the sampling rate.

It is then calculated:

K=1/(1+(G*SQRT(2))+G2)   (2)

In the formula (2), SQRT represents square root; and G is equal to G inthe formula (1).

It is next calculated:

B0=(1−(G*SQRT(2))+G2)*K   (3)

In the formula (3), G is equal to G in the formula (1); SQRT representssquare root; and K is equal to K in the formula (2).

It is then calculated:

B1=2*(G2−1)*K   (4)

In the formula (4), G is equal to G in the formula (1), and K is equalto K in the formula (1).

Then, B1 is assigned to A1, i.e., A1=B1, and next, B0 is assigned to A2,i.e., A2=B0.

The above-mentioned a0, a1, a2, b0, b1, and b2 may be equal to 1, A1,A2, B0, B1, and 1 respectively.

In this way, the non-normalized coefficients of the all-pass biquad IIRfiltering algorithm may be obtained, and may be used in the course ofperforming phase recovery.

It should be noted that the function implemented by the biquad IIRfiltering algorithm is the same as the function implemented by thebiquad IIR filter. The biquad IIR filter is a commonly used IIR filter.

In step 107, the high-frequency signals subjected to phase recovery andthe low-frequency signals are superimposed to obtain sampled audiosignal in which the high-frequency signals are recovered.

During the practice, the terminal may superimpose the high-frequencysignals subjected to phase recovery and the low-frequency signals toobtain sampled audio signal in which the high-frequency signals arerecovered.

Optionally, in step 103, if the first frequency point is not present,the following processing may be performed:

according to the FFT result, if the first frequency point is notpresent, converting the audio signal sampled at the preset number ofsampling points into a plurality of frequency subbands having an equalwidth, and synthesizing the audio signals of the plurality of frequencysubbands; separating the audio signal obtained by synthesizing the audiosignals of the plurality of frequency subbands according to a presetthird frequency point to obtain high-frequency signals and low-frequencysignals; and superimposing the high-frequency signals and thelow-frequency signals according to the preset third frequency point toobtain sampled audio signal.

The third frequency point may be a preset frequency point, and may bestored in the terminal, or may be a first frequency point determinedbased on audio signal sampled at a preset number of sampling points,which are buffered previously. For example, the audio signal sampled atthe preset number of sampling points are available currently, which arebuffered for the third time, the first frequency point may be determinedbased on the audio signal sampled at the preset number of samplingpoints, which are buffered for the second time.

During the practice, after obtaining the FFT result, if the FFT resultis a frequency spectrum, the terminal may calculate a power spectrumaccording to the frequency spectrum. In the power spectrum, eachfrequency point corresponds to one power. The terminal may then scan thepower spectrum to find a cliff-like attenuation point of power, that is,to find a first frequency point satisfying the preset conditions. If nofirst frequency point satisfying the preset conditions is present, theaudio signal sampled at the preset number of sampling points may beinput into an MDCT algorithm, and converted into audio signals of aplurality of frequency subbands having an equal width. Since the firstfrequency point is not present, the audio signals of the plurality offrequency subbands having an equal width may be input into an inverseMDCT algorithm to be synthesized, and the synthesized audio signal areobtained.

Then, the synthesized audio signal are subjected to linear high-passfiltering to obtain high-frequency signals, wherein the frequency ofeach of the high-frequency signals is greater than or equal to thefrequency of the third frequency point. In addition, the synthesizedaudio signal are subjected to linear low-pass filtering to obtainlow-frequency signals, wherein the frequency of each of thelow-frequency signals is less than the frequency of the third frequencypoint.

The low-frequency signals and the high-frequency signals may then besuperimposed to obtain the sampled audio signal.

Although the first frequency point is not present this time, in order toprevent a sudden change of the audio signals obtained by sampling forsuccessive two times, the frequency subbands are separated first, andthen subjected to synthesis and other processes.

It should be noted that, in the above process, for a compressed audio,the processing of the above steps 101 to 107 is performed every time theaudio signals of a preset number of sampling points are sampled, untilthe entire compressed audio has been recovered.

It should be noted that the audio in the embodiment of the presentdisclosure may be any audio format, such as MP3, AAC (Advanced AudioCoding, WMA (Windows Media Audio)), or the like. In addition, in thepresent disclosure, the data amount of the audio signal which isprocessed at a time is adjusted by adjusting the preset number in thestep 101, so as to be applicable to platforms having differentcalculation powers, and platforms having ultralow power consumption andweak computing power.

In an embodiment of the present disclosure, in case of an audio with alossy format, after audio signal sampled at a preset number of samplingpoints are buffered each time, the sampled audio signal are subjected toFFT to obtain an FFT result. According to the FFT result, if a firstfrequency point satisfying preset conditions is present, the audiosignal sampled at the preset number of sampling points are convertedinto audio signals of a plurality of frequency subbands having an equalwidth. A target frequency subband including the first frequency point isdetermined. Then, based on the audio signal of a previous frequencysubband of the target frequency subband, the audio signal of the targetfrequency subband in the plurality of frequency subbands and the audiosignals of the frequency subbands after the target frequency subband arerecovered. Next, the audio signals of the frequency subbands before thetarget frequency subband, the audio signal of the target frequencysubband, and the audio signals of the frequency subbands after thetarget frequency subband in the plurality of frequency subbands aresynthesized. The synthesized audio signal are separated according to thefirst frequency point to obtain high-frequency signals and low-frequencysignals, and the high-frequency signals are subjected to phase recovery.The high-frequency signals subjected to phase recovery and thelow-frequency signals are superimposed to obtain sampled audio signal inwhich the high-frequency signals are recovered. As such, since thehigh-frequency signals in the sampled audio signal may be recovered, thesampled audio signal are recovered as well. Therefore, the method forrecovering audio signals is provided.

Based on the same technical concept, an embodiment of the presentdisclosure further provides an apparatus for recovering audio signals.As shown in FIG. 3, the apparatus includes:

a buffering module 310, configured to buffer an audio signal sampled ata preset number of sampling points;

a FFT module 320, configured to perform FFT on the sampled audio signalto obtain an FFT result;

a converting module 330, configured to, according to the FFT result, ifa first frequency point satisfying preset conditions is present, convertthe audio signal sampled at the preset number of sampling points intoaudio signals of a plurality of frequency subbands having an equalwidth;

a determining module 340, configured to determine a target frequencysubband to which the first frequency point belongs, wherein the presetconditions are that a difference between frequencies of the firstfrequency point and a second frequency point is less than a first presetvalue, a difference between powers of the first frequency point and thesecond frequency point is greater than a second preset value, a power ofa frequency point having a frequency greater than the frequency of thefirst frequency point is zero, and the frequency of the second frequencypoint is less than the frequency of the first frequency point;

a recovering module 350, configured to, according to the audio signal ofa previous frequency subband of the target frequency subband, the audiosignal of the target frequency subband in the plurality of frequencysubbands and the audio signals of the frequency subbands after thetarget frequency subband;

a synthesizing module 360, configured to synthesize the audio signals ofthe frequency subbands before the target frequency subband in theplurality of frequency subbands, the audio signal of the targetfrequency subband, and the audio signals of the frequency subbands afterthe target frequency subband in the plurality of frequency subbands;

a separating module 370, configured to separate the synthesized audiosignal according to the first frequency point to obtain high-frequencysignals and low-frequency signals, wherein the recovering module 350 isfurther configured to perform phase recovery on the high-frequencysignals; and

a superimposiing module 380, configured to superimpose thehigh-frequency signals subjected to phase recovery and the low-frequencysignals to obtain sampled audio signal in which the high-frequencysignals are restored.

Optionally, the converting module 330 is further configured to,according to the FFT result, if the first frequency point is notpresent, convert the audio signal sampled at the preset number ofsampling points into a plurality of frequency subbands having an equalwidth;

the synthesizing module 360 is further configured to synthesize theaudio signals of the plurality of frequency subbands;

the separating module 370 is further configured to separate the audiosignal obtained by synthesizing the audio signals of the plurality offrequency subbands according to a preset third frequency point to obtainhigh-frequency signals and low-frequency signals; and

the superimposing module 380 is further configured to superimpose thehigh-frequency signals and the low-frequency signals according to thepreset third frequency point to obtain the sampled audio signal.

Optionally, the separating module 370 is configured to:

perform linear high-pass filtering on the synthesized audio signal toobtain the high-frequency signals, and perform linear low-pass filteringon the synthesized audio signal to obtain the low-frequency signals,wherein the frequency of each of the signals subjected to linearhigh-pass filtering is greater than or equal to the frequency of thefirst frequency point, and the frequency of each of the signalssubjected to linear low-pass filtering is less than the frequency of thefirst frequency point.

Optionally, the recovering module 350 is configured to:

perform all-pass biquad IIR filtering on the high-frequency signals toobtain high-frequency signals subjected to phase recovery.

Optionally, the determining module 340 is further configured to:

determine a coefficient of the biquad IIR filtering according to thefrequency of the first frequency point and sampling rates.

Optionally, as shown in FIG. 4, the apparatus further includes:

a windowing module 390 configured to, prior to the performing FFT on thesampled audio signal to obtain an FFT result, window the sampled audiosignal to obtain audio signal subjected to windowing; and

the FFT module 320 is configured to:

perform the FFT on the audio signal subjected to windowing to obtain theFFT result.

In an embodiment of the present disclosure, in case of an audio with alossy format, after audio signal sampled at a preset number of samplingpoints is buffered each time, the sampled audio signal are subjected toFFT to obtain an FFT result. According to the FFT result, if a firstfrequency point satisfying preset conditions is present, the audiosignal sampled at the preset number of sampling points are convertedinto audio signals of a plurality of frequency subbands having an equalwidth. A target frequency subband including the first frequency point isdetermined. Then, based on the audio signal of a previous frequencysubband of the target frequency subband, the audio signal of the targetfrequency subband in the plurality of frequency subbands and the audiosignals of the frequency subbands after the target frequency subband arerecovered. Next, the audio signals of the frequency subbands before thetarget frequency subband, the audio signal of the target frequencysubband, and the audio signals of the frequency subbands after thetarget frequency subband in the plurality of frequency subbands aresynthesized. The synthesized audio signal are separated according to thefirst frequency point to obtain high-frequency signals and low-frequencysignals, and the high-frequency signals are subjected to phase recovery.The high-frequency signals subjected to phase recovery and thelow-frequency signals are superimposed to obtain sampled audio signal inwhich the high-frequency signals are recovered. As such, since thehigh-frequency signals in the sampled audio signal may be recovered, thesampled audio signal are recovered as well. Therefore, the method forrecovering audio signals is provided.

It should be noted that, when recovering audio signals, the apparatusfor recovering audio signals is only illustrated by taking division ofthe all functional module as an example. While in a practicalapplication, the above functions may be assigned to different modules tobe achieved according to needs. That is, an internal structure of theterminal may be divided into the different functional modules, so as toachieve all or part of the functions described above.In addition, theapparatus for live broadcasting and the method for live broadcastingprovided by the forging embodiments belong to the same concept. Specificimplementation processes of the apparatus may refer to the embodimentsof the method, and details thereof will not be repeated herein.

FIG. 5 is a structural block diagram of a terminal 500 according to anexemplary embodiment of the present disclosure. The terminal 500 may bea smart phone, a tablet computer, a Moving Picture Experts Group AudioLayer III (MP3) player, a Moving Picture Experts Group Audio Layer IV(MP4) player, or a laptop or desktop computer. The terminal 500 may alsobe referred to as a user equipment, a portable terminal, a laptopterminal, a desktop terminal, or the like

Generally, the terminal 500 includes a processor 501 and a memory 502.

The processor 501 may include one or more processing cores, such as a4-core processor, an 8-core processor, or the like. The processor 501may be practiced by using at least one of hardware forms in a digitalsignal processor (DSP), a field-programmable gate array (FPGA) and aprogrammable logic array (PLA). The processor 501 may also include amain processor and a co-processor. The main processor is a processor forprocessing data in an awaken state, and is also called as a centralprocessing unit (CPU). The co-processor is a low-power processor forprocessing data in a standby state. In some embodiments, the processor501 may be integrated with a graphics processing unit (GPU) which isresponsible for rendering and drawing of content required to bedisplayed by a display. In some embodiments, the processor 501 may alsoinclude an artificial intelligence (AI) processor for processing acalculation operation related to machine learning.

The memory 502 may include one or more computer-readable storage mediawhich may be non-transitory. The memory 502may also include a high-speedrandom-access memory, as well as a non-volatile memory, such as one ormore disk storage devices and flash storage devices. In someembodiments, the non-transitory computer-readable storage medium in thememory 502 is configured to store at least one instruction which isexecutable by the processor 501 to implement the method for determiningthe karaoke singing score according to the embodiments of the presentdisclosure.

In some embodiments, the terminal 500 may optionally include aperipheral device interface 503 and at least one peripheral device. Theprocessor 501, the memory 502 and the peripheral device interface 503may be connected to each other via a bus or a signal line. The at leastone peripheral device may be connected to the peripheral deviceinterface 503 via a bus, a signal line or a circuit board. Specifically,the peripheral device includes at least one of a radio frequency circuit504, a touch display screen 505, a camera assembly 506, an audio circuit507, a positioning assembly 508 and a power source 509.

The peripheral device interface 503 may be configured to connect the atleast one peripheral device related to input/output (I/O) to theprocessor 501 and the memory 502. In some embodiments, the processor501, the memory 502 and the peripheral device interface 503 areintegrated on the same chip or circuit board. In some other embodiments,any one or two of the processor 501, the memory 502 and the peripheraldevice interface 503 may be practiced on a separate chip or circuitboard, which is not limited in this embodiment.

The radio frequency circuit 504 is configured to receive and transmit aradio frequency (RF) signal, which is also referred to as anelectromagnetic signal. The radio frequency circuit 504 communicateswith a communication network or another communication device via theelectromagnetic signal. The radio frequency circuit 504 converts anelectrical signal to an electromagnetic signal and sends the signal, orconverts a received electromagnetic signal to an electrical signal.Optionally, the radio frequency circuit 504 includes an antenna system,an RF transceiver, one or a plurality of amplifiers, a tuner, anoscillator, a digital signal processor, a codec chip set, a subscriberidentification module card or the like. The radio frequency circuit 504may communicate with another terminal based on a wireless communicationprotocol. The wireless communication protocol includes, but not limitedto: a metropolitan area network, generations of mobile communicationnetworks (including 2G, 3G, 4G and 5G), a wireless local area networkand/or a wireless fidelity (WiFi) network. In some embodiments, theradio frequency circuit 504 may further include a near fieldcommunication (NFC)-related circuits, which is not limited in thepresent disclosure.

The display screen 505 may be configured to display a user interface(UI). The UE may include graphics, texts, icons, videos and anycombination thereof. When the display screen 505 is a touch displayscreen, the display screen 505 may further have the capability ofacquiring a touch signal on a surface of the display screen 505 or abovethe surface of the display screen 505. The touch signal may be input tothe processor 501 as a control signal, and further processed therein. Inthis case, the display screen 505 may be further configured to provide avirtual button and/or a virtual keyboard or keypad, also referred to asa soft button and/or a soft keyboard or keypad. In some embodiments, onedisplay screen 505 may be provided, which is arranged on a front panelof the terminal 500. In some other embodiments, at least two displayscreens 505 are provided, which are respectively arranged on differentsurfaces of the terminal 500 or designed in a folded fashion. In stillsome other embodiments, the display screen 505 may be a flexible displayscreen, which is arranged on a bent surface or a folded surface of theterminal 500. Even, the display screen 505 may be further arranged to anirregular pattern which is non-rectangular, that is, a specially-shapedscreen. The display screen 505 may be fabricated from such materials asa liquid crystal display (LCD), an organic light-emitting diode (OLED)and the like.

The camera assembly 506 is configured to capture an image or a video.Optionally, the camera assembly 506 includes a front camera and a rearcamera. Generally, the front camera is arranged on a front panel of theterminal, and the rear camera is arranged on a rear panel of theterminal. In some embodiments, at least two rear cameras are arranged,which are respectively any one of a primary camera, a depth of field(DOF) camera, a wide-angle camera and a long-focus camera, such that theprimary camera and the DOF camera are fused to implement the backgroundvirtualization function, and the primary camera and the wide-anglecamera are fused to implement the panorama photographing and virtualreality (VR) photographing functions or other fused photographingfunctions. In some embodiments, the camera assembly 506 may furtherinclude a flash. The flash may be a single-color temperature flash or adouble-color temperature flash. The double-color temperature flashrefers to a combination of a warm-light flash and a cold-light flash,which may be used for light compensation under different colortemperatures.

The audio circuit 507 may include a microphone and a speaker. Themicrophone is configured to capture an acoustic wave of a user and anenvironment, and convert the acoustic wave to an electrical signal andoutput the electrical signal to the processor 501 for furtherprocessing, or output to the radio frequency circuit 504 to implementvoice communication. For the purpose of stereo capture or noisereduction, a plurality of such microphones may be provided, which arerespectively arranged at different positions of the terminal 500. Themicrophone may also be a microphone array or an omnidirectionalcapturing microphone. The speaker is configured to convert an electricalsignal from the processor 501 or the radio frequency circuit 504 to anacoustic wave. The speaker may be a traditional thin-film speaker, ormay be a piezoelectric ceramic speaker. When the speaker is apiezoelectric ceramic speaker, an electrical signal may be converted toan acoustic wave audible by human beings, or an electrical signal may beconverted to an acoustic wave inaudible by human beings for the purposeof ranging or the like. In some embodiments, the audio circuit 507 mayfurther include a headphone plug.

The positioning assembly 508 is configured to determine a currentgeographical position of the terminal 500 to implement navigation or alocal based service (LBS). The positioning assembly 508 may be theglobal positioning system (GPS) from the United States, the Beidoupositioning system from China, the Grenas satellite positioning systemfrom Russia or the Galileo satellite navigation system from the EuropeanUnion.

The power source 509 is configured to supply power for the components inthe terminal 500. The power source 509 may be an alternating current, adirect current, a disposable battery or a rechargeable battery. When thepower source 509 includes a rechargeable battery, the rechargeablebattery may support wired charging or wireless charging. Therechargeable battery may also support the supercharging technology.

In some embodiments, the terminal may further include one or a pluralityof sensors 510. The one or plurality of sensors 510 include, but notlimited to: an acceleration sensor 511, a gyroscope sensor 512, apressure sensor 513, a fingerprint sensor 514, an optical sensor 515 anda proximity sensor 516.

The acceleration sensor 511 may detect accelerations on three coordinateaxes in a coordinate system established for the terminal 500. Forexample, the acceleration sensor 511 may be configured to detectcomponents of a gravity acceleration on the three coordinate axes. Theprocessor 501 may control the touch display screen 505 to display theuser interface in a horizontal view or a longitudinal view based on agravity acceleration signal acquired by the acceleration sensor 511. Theacceleration sensor 511 may be further configured to acquire motion dataof a game or a user.

The gyroscope sensor 512 may detect a direction and a rotation angle ofthe terminal 500, and the gyroscope sensor 512 may collaborate with theacceleration sensor 511 to capture a 3D action performed by the user forthe terminal 500. Based on the data acquired by the gyroscope sensor512, the processor 501 may implement the following functions: actionsensing (for example, modifying the UE based on an inclination operationof the user), image stabilization during the photographing, game controland inertial navigation.

The force sensor 513 may be arranged on a side frame of the terminaland/or on a lowermost layer of the touch display screen 505. When theforce sensor 513 is arranged on the side frame of the terminal 500, agrip signal of the user against the terminal 500 may be detected, andthe processor 501 implements left or right hand identification orperform a shortcut operation based on the grip signal acquired by theforce sensor 513. When the force sensor 513 is arranged on the lowermostlayer of the touch display screen 505, the processor 501 implementcontrol of an operable control on the UI based on a force operation ofthe user against the touch display screen 505. The operable controlincludes at least one of a button control, a scroll bar control, an iconcontrol, and a menu control.

The fingerprint sensor 514 is configured to acquire fingerprints of theuser, and the processor 501 determines the identity of the user based onthe fingerprints acquired by the fingerprint sensor 514, or thefingerprint sensor 514 determines the identity of the user based on theacquired fingerprints. When it is determined that the identify of theuser is trustable, the processor 501 authorizes the user to performrelated sensitive operations, wherein the sensitive operations includeunlocking the screen, checking encrypted information, downloadingsoftware, paying and modifying settings and the like. The fingerprintsensor 514 may be arranged on a front face a back face or a side face ofthe terminal 500. When the terminal 500 is provided with a physical keyor a manufacturer's logo, the fingerprint sensor 514 may be integratedwith the physical key or the manufacturer's logo.

The optical sensor 515 is configured to acquire the intensity of ambientlight. In one embodiment, the processor 501 may control a displayluminance of the touch display screen 505 based on the intensity ofambient light acquired by the optical sensor 515. Specifically, when theintensity of ambient light is high, the display luminance of the touchdisplay screen 505 is up-shifted; and when the intensity of ambientlight is low, the display luminance of the touch display screen 505 isdown-shifted. In another embodiment, the processor 501 may furtherdynamically adjust photographing parameters of the camera assembly 506based on the intensity of ambient light acquired by the optical sensor.

The proximity sensor 516, also referred to as a distance sensor, isgenerally arranged on the front panel of the terminal 500. The proximitysensor 516 is configured to acquire a distance between the user and thefront face of the terminal 500. In one embodiment, when the proximitysensor 516 detects that the distance between the user and the front faceof the terminal 500 gradually decreases, the processor 501 controls thetouch display screen 505 to switch from an active state to a rest state;and when the proximity sensor 516 detects that the distance between theuser and the front face of the terminal 500 gradually increases, theprocessor 501 controls the touch display screen 505 to switch from therest state to the active state.

A person skilled in the art may understand that the structure of theterminal as illustrated in FIG. 5 does not construe a limitation on theterminal 500. The terminal may include more components over thoseillustrated in FIG. 5, or combinations of some components, or employdifferent component deployments.

Persons of ordinary skill in the art can understand that all or part ofthe steps described in the above embodiments can be completed throughhardware, or through relevant hardware instructed by applications storedin a non-transitory computer readable storage medium, such as aread-only memory, a disk or a CD.

Described above are merely exemplary embodiments of the presentdisclosure, and are not intended to limit the present disclosure. Withinthe spirit and principles of the disclosure, any modifications,equivalent substitutions or improvements are within the protection scopeof the present disclosure.

1. A method for recovering audio signals, comprising: determining afirst frequency point in an audio signal having a high-frequency signalto be recovered by power spectrum scanning, wherein the first frequencypoint is a frequency point having a minimum frequency of thehigh-frequency signal to be recovered in the audio signal; among aplurality of frequency subbands having equal width of the audio signalhaving the high-frequency signal to be recovered, recovering, accordingto the audio signal of a previous frequency subband of a targetfrequency subband, the audio signal of the target frequency subband andthe audio signals of the frequency subbands after the target frequencysubband in the plurality of frequency subbands, wherein the targetfrequency subband is a frequency subband to which the first frequencypoint belongs; synthesizing the audio signals of the frequency subbandsbefore the target frequency subband in the plurality of frequencysubbands, the audio signal of the target frequency subband, and theaudio signals of the frequency subbands after the target frequencysubband in the plurality of frequency subbands; separating thesynthesized audio signal according to the first frequency point toobtain high-frequency signals and low-frequency signals, and performingphase recovery on the high-frequency signals; and superimposing thehigh-frequency signals subjected to phase recovery and the low-frequencysignals to obtain an audio signal in which the high-frequency signalsare recovered.
 2. The method according to claim 1, further comprising:if the first frequency point is not present in the audio signal having ahigh-frequency signal to be recovered, converting the audio signalhaving a high-frequency signal to be recovered into a plurality offrequency subbands having an equal width, and synthesizing the audiosignals of the plurality of frequency subbands; separating the audiosignal obtained by synthesizing the audio signals of the plurality offrequency subbands according to a preset third frequency point to obtainhigh-frequency signals and low-frequency signals; and superimposing thehigh-frequency signals and the low-frequency signals obtained byseparating according to the preset third frequency point to obtain theaudio signal in which the high-frequency signals are recovered.
 3. Themethod according to claim 1, wherein separating the synthesized audiosignal according to the first frequency point to obtain high-frequencysignals and low-frequency signals comprises: performing linear high-passfiltering on the synthesized audio signal to obtain the high-frequencysignals, and performing linear low-pass filtering on the synthesizedaudio signal to obtain the low-frequency signals, wherein a frequency ofeach of the signals subjected to linear high-pass filtering is greaterthan or equal to the frequency of the first frequency point, and afrequency of each of the signals subjected to linear low-pass filteringis less than the frequency of the first frequency point.
 4. The methodaccording to claim 1, wherein performing phase recovery on thehigh-frequency signals comprises: performing all-pass biquad infiniteimpulse response (IIR) filtering on the high-frequency signals to obtainthe high-frequency signals subjected to phase recovery.
 5. The methodaccording to claim 4, further comprising: determining a coefficient ofthe biquad IIR filtering according to the frequency of the firstfrequency point and sampling rates.
 6. The method according to claim 13,wherein prior to the performing FFT on the sampled audio signal toobtain an FF1 result, the method further comprises: windowing thesampled audio signal to obtain audio signal subjected to windowing; andwherein performing FFT on the sampled audio signal to obtain an FFTresult comprises: performing the FFT on the audio signal subjected towindowing to obtain the FFT result.
 7. (canceled)
 8. (canceled) 9.(canceled)
 10. (canceled)
 11. (canceled)
 12. (canceled)
 13. The methodaccording to claim 1, wherein determining a first frequency point in anaudio signal having a high-frequency signal to be recovered by powerspectrum scanning comprises: buffering an audio signal which is sampledat a preset number of sampling points; performing fast Fourier transform(FFT) on the sampled audio signal to obtain an EFT result; according tothe FFT result, finding a first frequency point that satisfying presetconditions, wherein the preset conditions are that a difference betweenfrequencies of the first frequency point and a second frequency point isless than a first preset value, a difference between powers of the firstfrequency point and the second frequency point is greater than a secondpreset value, a power of a frequency point having a frequency greaterthan the frequency of the first frequency point is zero, and thefrequency of the second frequency point is less than the frequency ofthe first frequency point.
 14. A terminal, comprising a memory and aprocessor, wherein the memory is used to store at least one instruction,the processor is used to implement a method when executing the at leastone instruction, and the method comprises: determining a first frequencypoint in an audio signal having a high-frequency signal to be recoveredby power spectrum scanning, wherein the first frequency point is afrequency point having a minimum frequency of the high-frequency signalto be recovered in the audio signal; among a plurality of frequencysubbands having equal width of the audio signal having thehigh-frequency signal to be recovered, recovering, according to theaudio signal of a previous frequency subband of a target frequencysubband, the audio signal of the target frequency subband and the audiosignals of the frequency subbands after the target frequency subband inthe plurality of frequency subbands, wherein the target frequencysubband is a frequency subband to which the first frequency pointbelongs: synthesizing the audio signals of the frequency subbands beforethe target frequency subband in the plurality of frequency subbands, theaudio signal of the target frequency subband, and the audio signals ofthe frequency subbands after the target frequency subband in theplurality of frequency subbands; separating the synthesized audio signalaccording to the first frequency point to obtain high-frequency signalsand low-frequency signals, and performing phase recovery on thehigh-frequency signals; and superimposing the high-frequency signalssubjected to phase recovery and the low-frequency signals to obtain anaudio signal in which the high-frequency signals are recovered.
 15. Theterminal according to claim 14, wherein the method further comprises: ifthe first frequency point is not present in the audio signal having ahigh-frequency signal to be recovered, converting the audio signalhaving a high-frequency signal to be recovered into a plurality offrequency subbands having an equal width, and synthesizing the audiosignals of the plurality of frequency subbands; separating the audiosignal obtained by synthesizing the audio signals of the plurality offrequency subbands according to a preset third frequency point to obtainhigh-frequency signals and low-frequency signals; and superimposing thehigh-frequency signals and the low-frequency signals obtained byseparating according to the preset third frequency point to obtain theaudio signal in which the high-frequency signals are recovered.
 16. Theterminal according to claim 14, wherein separating the synthesized audiosignal according to the first frequency point to obtain high-frequencysignals and low-frequency signals comprises: performing linear high-passfiltering on the synthesized audio signal to obtain the high-frequencysignals, and performing linear low-pass filtering on the synthesizedaudio signal to obtain the low-frequency signals, wherein a frequency ofeach of the signals subjected to linear high-pass filtering is greaterthan or equal to the frequency of the first frequency point, and afrequency of each of the signals subjected to linear low-pass filteringis less than the frequency of the first frequency point.
 17. Theterminal according to claim 14, wherein performing phase recovery on thehigh-frequency signals comprises: performing all-pass biquad infiniteimpulse response (IIR) filtering on the high-frequency signals to obtainthe high-frequency signals subjected to phase recovery.
 18. The terminalaccording to claim 17, wherein the method further comprises: determininga coefficient of the biquad IIR filtering according to the frequency ofthe first frequency point and sampling rates.
 19. The terminal accordingto claim 14, wherein determining a first frequency point in an audiosignal having a high-frequency signal to be recovered by power spectrumscanning comprises: buffering an audio signal which is sampled at apreset number of sampling points; performing fast Fourier transform(FFT) on the sampled audio signal to obtain an FFT result; according tothe FFT result, finding a first frequency point that satisfying presetconditions, wherein the preset conditions are that a difference betweenfrequencies of the first frequency point and a second frequency point isless than a first preset value, a difference between powers of the firstfrequency point and the second frequency point is greater than a secondpreset value, a power of a frequency point having a frequency greaterthan the frequency of the first frequency point is zero, and thefrequency of the second frequency point is less than the frequency ofthe first frequency point.
 20. The terminal according to claim 19,wherein prior to the performing FFT on the sampled audio signal toobtain an FFT result, the method further comprises: windowing thesampled audio signal to obtain audio signal subjected to windowing; andwherein performing FFT on the sampled audio signal to obtain an FFTresult comprises: performing the FFT on the audio signal subjected towindowing to obtain the FFT result.
 21. A non-transitorycomputer-readable storage medium storing at least one instruction,wherein the at least one instruction, when being executed by aprocessor, implements a method, and the method comprises: determining afirst frequency point in an audio signal having a high-frequency signalto be recovered by power spectrum scanning, wherein the first frequencypoint is a frequency point having a minimum frequency of thehigh-frequency signal to be recovered in the audio signal; among aplurality of frequency subbands having equal width of the audio signalhaving the high-frequency signal to be recovered, recovering, accordingto the audio signal of a previous frequency subband of a targetfrequency subband, the audio signal of the target frequency subband andthe audio signals of the frequency subbands after the target frequencysubband in the plurality of frequency subbands, wherein the targetfrequency subband is a frequency subband to which the first frequencypoint belongs; synthesizing the audio signals of the frequency subbandsbefore the target frequency subband in the plurality of frequencysubbands, the audio signal of the target frequency subband, and theaudio signals of the frequency subbands after the target frequencysubband in the plurality of frequency subbands; separating thesynthesized audio signal according to the first frequency point toobtain high-frequency signals and low-frequency signals, and performingphase recovery on the high-frequency signals; and superimposing thehigh-frequency signals subjected to phase recovery and the low-frequencysignals to obtain an audio signal in which the high-frequency signalsare recovered.
 22. The storage medium according to claim 21, wherein themethod further comprises: if the first frequency point is not present inthe audio signal having a high-frequency signal to be recovered,converting the audio signal having a high-frequency signal to berecovered into a plurality of frequency subbands having an equal width,and synthesizing the audio signals of the plurality of frequencysubbands; separating the audio signal obtained by synthesizing the audiosignals of the plurality of frequency subbands according to a presetthird frequency point to obtain high-frequency signals and low-frequencysignals; and superimposing the high-frequency signals and thelow-frequency signals obtained by separating according to the presetthird frequency point to obtain the audio signal in which thehigh-frequency signals are recovered.
 23. The storage medium accordingto claim 21, wherein separating the synthesized audio signal accordingto the first frequency point to obtain high-frequency signals andlow-frequency signals comprises: performing linear high-pass filteringon the synthesized audio signal to obtain the high-frequency signals,and performing linear low-pass filtering on the synthesized audio signalto obtain the low-frequency signals, wherein a frequency of each of thesignals subjected to linear high-pass filtering is greater than or equalto the frequency of the first frequency point, and a frequency of eachof the signals subjected to linear low-pass filtering is less than thefrequency of the first frequency point.
 24. The storage medium accordingto claim 21, wherein performing phase recovery on the high-frequencysignals comprises: performing all-pass biquad infinite impulse response(IIR) filtering on the high-frequency signals to obtain thehigh-frequency signals subjected to phase recovery.
 25. The storagemedium according to claim 24, wherein the method further comprises:determining a coefficient of the biquad IIR filtering according to thefrequency of the first frequency point and sampling rates.
 26. Thestorage medium according to claim 21, wherein determining a firstfrequency point in an audio signal having a high-frequency signal to berecovered by power spectrum scanning comprises: buffering an audiosignal which is sampled at a preset number of sampling points;performing fast Fourier transform (FFT) on the sampled audio signal toobtain an FFT result; according to the FFT result, finding a firstfrequency point that satisfying preset conditions, wherein the presetconditions are that a difference between frequencies of the firstfrequency point and a second frequency point is less than a first presetvalue, a difference between powers of the first frequency point and thesecond frequency point is greater than a second preset value, a power ofa frequency point having a frequency greater than the frequency of thefirst frequency point is zero, and the frequency of the second frequencypoint is less than the frequency of the first frequency point.